US 7295549 B2
A coding system and method for a terminal including a multi-rate codec is disclosed. The terminal includes a multi-rate adaptive coder that is capable of transmitting a continuous voice stream transmission at a source code bit rate and a channel code bit rate. A quality of service probing module probes an end-to-end network path of the continuous voice stream transmission to obtain at least one quality of service parameter. A quality of service management module determines at least one constraint associated with the continuous voice stream transmission. An adaptive bit rate algorithm module dynamically adjusts the source code bit rate and the channel code bit rate as a function of the quality of service parameter and the constraint to obtain a maximum value of perceived user performance during the continuous voice stream transmission.
1. A terminal having an adaptive coding system, comprising:
a multi-rate adaptive coder capable of transmitting a continuous voice stream transmission at a source code bit rate and a channel code bit rate;
a quality of service probing module for probing an end-to-end network path of the continuous voice stream transmission to obtain at least one quality of service parameter;
a quality of service management module for determining at least one constraint associated with the continuous voice stream transmission; and
an adaptive bit rate algorithm module for dynamically adjusting the source code bit rate and the channel code bit rate as a function of the quality of service parameter and the constraint to obtain a maximum value of perceived user performance during the continuous voice stream transmission.
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19. A method of providing adaptive coder rates to a terminal using an access network, comprising the steps of:
providing a multi-rate adaptive coder capable of transmitting a continuous voice stream transmission that has a source code bit rate and a channel code bit rate;
probing an end-to-end network path of the voice stream transmission to obtain at least one quality of service parameter;
determining at least one constraint associated with the continuous voice stream transmission; and
adapting the source code bit rate and the channel code bit rate as a function of the quality of service parameter and the constraint to obtain a maximum value of perceived user performance during the continuous voice stream transmission.
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40. A terminal having an adaptive coding system, comprising:
means for transmitting a continuous voice stream transmission that includes a source code bit rate and a channel code bit rate;
means for probing an end-to-end network path of the continuous voice stream transmission to obtain at least one quality of service parameter;
means for determining at least one constraint associated with the continuous voice stream transmission; and
means for adjusting the source code bit rate and the channel code bit rate as a function of the quality of service parameter and the constraint to obtain a maximum value of perceived user performance during the continuous voice stream transmission.
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44. A method of jointly adapting a codec source rate and a forward error correction (FEC) channel rate to optimize perceived user quality, comprising the steps of:
probing an end-to-end path to identify at least one quality of service (QoS) parameter for a voice stream transmission;
querying a quality of service (QoS) manager to identify at least one quality of service (QoS) constraint for the voice stream transmission;
processing the quality of service (QoS) parameter to produce at least one QoS estimate for the voice stream transmission;
processing the quality of service constraint (QoS) to produce at least one QoS constraint on a selection of source and channel code bit rates.
exploring different combinations of codec source rates and forward error correction (FEC) channel rates to produce at least one speech quality estimate; and
choosing at least one combination of a respective codec source rate and a respective forward error correction (FEC) channel rate that maximizes perceived user quality.
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1. Technical Field
The present invention relates generally to source and channel coding for speech codecs and more particularly, to a method and system for providing an optimal choice of source and channel code bit rates given information on the packet loss and available bandwidth of a IP access network.
2. Related Art
Existing speech coders were not designed for use over IP packet networks. A packet switched network is a shared medium designed for asynchronous transmission on a best effort basis. In IP networks the available bandwidth and delay vary over time. Time-critical applications such as voice and video have traditionally assumed guaranteed bandwidth, delay and synchronous transmission. Most of the speech coders operate under preset schemes for data and channel code rates making them vulnerable to the varying conditions on wired and wireless IP-based hops. Some kind of adaptation is therefore needed to dynamically adapt the codec bit rate to quickly changing network conditions to preserve acceptable levels of reliability and quality.
Several types of degradations occur in IP networks, among them: 1) packet loss due to network congestion resulting from a lack of available bandwidth, 2) packet loss due to the network jitter, 3) delay due to packetization, 4) delay due to congestion, and 5) packet loss due to random or bursty communication noise. The first four degradations are prominent in the wired IP networks while the last one occurs primarily in wireless networks and is due to residual bit errors at the link layer. The retransmission mechanism used in the TCP protocol for error control cannot be used due to its inherent delay that might be unacceptable for real-time, interactive voice applications.
Codec rate adaptation is an effective method to cope with network congestion and mitigate the effects of packet loss. The European Telecommunication Standards Institute/3rd Generation Partnership Project (“ETSI/3GPP”) adaptive multi-rate (AMR) speech codec is suitable for this purpose. In some systems, the packet loss is reduced by dividing the network conditions into eight states and assigning each state to one of the eight bit rates of the AMR codec. Network conditions are monitored using the difference in timestamps between successive speech frames at the receiving side (i.e., the system monitors the network jitter). The results show a drastic reduction in packet loss rate when compared to fixed-rate codecs such as G.711 and G.723.1.
In some situations, reducing the source bit rate alone does not help. Such situations may be short-term transient congestion, congestion caused by others' traffic or residual bit errors caused by a noisy wireless link. Channel coding—or forward error correction (FEC)—can then be used in conjunction with an error control scheme to optimally allocate the amount of redundant bits and information bits in response to varying channel conditions (available bandwidth, loss rate, delay, etc.)
Others have proposed a flexible scheme for voice transmission over the mobile Internet in which the AMR codec is combined with a systematic convolutional code of rate 1/n. The packetization scheme is done according to the optimal puncturing patterns of the code, that is, all bits stemming from the same generator polynomial are put into the same packet. Hence, one media packet is followed by n−1 forward error correction (FEC) packets. Packet loss is assumed as puncturing of the convolutional code for which decoding methods are known. The code reduces both random packet loss in the wired IP network and random bit errors on the wireless link (similar to a physical layer FEC code). Although the authors mention that appropriate feedback can adaptively control the amount of source and channel bits, they do not implement any rate adaptation scheme.
Another solution proposes using redundant audio coding to deliver decent audio quality to a destination. In this scheme, an audio packet includes the encoded main information as well as a highly compressed version of previous packets (the redundant information). For instance, packet n includes in addition to the PCM encoded samples LPC or GSM versions of packets n−1 and n−2. The perceived loss rate is used as the metric and gradually changes the amount of redundancy according to a fixed threshold (e.g., 3% packet loss rate). Loss rates are sent back to the encoders using RTCP reports. However, this algorithm suffers from several shortcomings, such as a cyclical behavior (increasing and decreasing the redundancy continuously although the network loss rate is near constant) that results in poor performance.
In a modification of the above-referenced approach, the problem is stated as a constrained optimization one: given the available bandwidth, what is the combination of main and redundant information which provides the best perceived audio quality? This approach still uses redundant audio coding, but also tries to optimize a subjective measure of quality taking into account the rate and delay constraints. In this approach, it is found that the main information should be encoded using the highest quality coder and copies located towards the end should be encoded with the next highest quality coders. The available bandwidth and loss rate are obtained using TCP-friendly rate control and RTCP feedback, respectively.
The need to distinguish network congestion from bit errors on radio links serves as a basis for a new quality of service (QoS) control architecture, in an article entitled “Rate and robustness control with RTP monitoring agent for mobile multimedia streaming,”. In this paper, a new type of proxy is introduced called “RTP monitoring agent” located at the edge of the wired network and wireless link. The RTP monitoring agent sends feedback reports about the wired network conditions such as jitter, loss, etc. to the media server. The latter also receives RTCP reports from the media receiver containing statistics about both the wired and the wireless networks. The media server is then able to apply the appropriate strategy depending on whether packet losses are due to network congestion (reduce the encoder bit rate) or radio link errors (increase robustness by adding more FEC). The adaptation algorithm consists of pre-defined combinations targeted to make the total packet loss rate less than 1%.
The present invention discloses a fixed or wireless terminal having adaptive coder bit rates that are capable of being adjusting during operation to achieve a maximum user perceived performance. The terminal or a network includes a multi-rate adaptive coder capable of transmitting a voice stream transmission that includes a source code bit rate and a channel code bit rate. A quality of service probing module is also included on the terminal that is operable to probe an end-to-end network path of the voice stream transmission to obtain at least one quality of service parameter. The terminal also includes a quality of service management module that is programmed to determine at least one constraint that is associated with the voice stream transmission. An adaptive bit rate algorithm module is programmed to jointly adjust the source code bit rate and the channel code bit rate as a function of the quality of service parameter and the constraint to obtain a maximum value of perceived user performance during the voice stream transmission.
The adaptive bit rate algorithm module computes a maximum number of media symbols (k) in a codeword, a maximum length of the codeword (n), an end-to-end network delay, a delay impairment factor, a measured packet loss factor, a signal distortion factor and an R-factor to compute a mean opinion score (MOS). The mean opinion score is used to adjust the source code bit rate and the channel code bit rate. The mean opinion score (MOS) is determined for a plurality of combinations of source and channel code bit rate values. The source code bit rate and the channel code bit rate that are chosen for the voice stream transmission are the ones that achieve the highest mean opinion score for a give source code bit rate.
The present invention also discloses a method of providing adaptive coder rates to a terminal. One preferred method comprises the steps of (a) providing a multi-rate adaptive coder capable of transmitting a voice stream transmission that includes a source code bit rate and a channel code bit rate; (b) probing an end-to-end network path of the voice stream transmission to obtain at least one quality of service parameter; (c) determining at least one constraint associated with the voice stream transmission; and (d) adapting the source code bit rate and the channel code bit rate as a function of the quality of service parameter and the constraint to obtain a maximum value of perceived user performance during the voice stream transmission.
Other systems, methods, features and advantages of the invention will be, or will become, apparent to one with skill in the art upon examination of the following figures and detailed description. It is intended that all such additional systems, methods, features and advantages be included within this description, be within the scope of the invention, and be protected by the following claims.
The invention can be better understood with reference to the following drawings and description. The components in the figures are not necessarily to scale, emphasis instead being placed upon illustrating the principles of the invention. Moreover, in the figures, like reference numerals designate corresponding parts throughout the different views.
Referring collectively to
The fixed access network 108 includes a server 116 that is connected with a respective router 114. The connection between the fixed terminal 106 and the fixed access network 108 may be referred to as a fixed hop. The routers 114 of the wireless access point 104 and the fixed access point 108 may be connected with a plurality of network routers 118 that establish a connection between the source nodes or terminals 102, 106 and either a wireless destination node or terminal 120 or a fixed destination node or terminal 122. As such, during operation a connection is established between the source terminals 102, 106 and the destination terminals 120, 122.
As further illustrated in
In the preferred embodiment of the present invention, the source terminals 102, 106 use an adaptive multi-rate (AMR) speech codec or coder 202 to transmit voice data to the destination terminals 120, 122. The AMR speech codec is a variable rate speech codec that may be used in W-CDMA communication. As set forth above, the AMR speech codec was developed by the ETSI and adopted by the 3GPP for use in VoIP communication.
The coding scheme used is the algebraic code excited linear prediction (ACELP) coder. Voice activity detection, comfort noise generation, source controlled rate operation and error concealment of lost frames are also provided in the specifications. Referring to
In the present invention, the three classes are subject to the same level of error protection. Preferentially, the real-time transport protocol (RTP) is used over the IP access network 100. RFC 3267 defines the RTP payload format for the AMR speech codec. The structure of the preferred RTP payload format is shown in
In the present invention, redundancy is spread across packets in the manner set forth below. Each media packet can be seen as consisting of several symbols, each symbol being an element of GF(28). A group of k symbols from k different media packets is used to generate n-k redundant symbols, creating a length n codeword. These redundant symbols are then placed in separate packets as shown in
To assess the tradeoff between source and channel bit or code rates (Rs and Rc, respectively), the present invention relates these and other QoS factors, such as packet loss and available bandwidth, to a criterion that can be used as a basis of comparison. One such criterion may be based on the Emodel. The Emodel is a computational model standardized by the ITU-T in 1998 that combines the effects of variations in several transmission parameters to predict the subjective conversational quality of 3.1 kHz handset telephony. The output of the Emodel is a scalar quantity rating value or R-Factor (R), on a scale of 0 to 100, where R=0 represents an extremely bad quality and R=100 represents a very high quality. The difference between a mean opinion score (MOS) and the R-Factor (R) scores is that the former is based on real subjective ratings of a large number of voice calls by a large set of people, whereas the latter is an analytical estimate of the subjective score.
The mean opinion score (MOS) is used to refer to the quality of a digitized voice line, as defined by the ITU-T P.800. The mean opinion score (MOS) is an average of the numbers for a particular codec. It is derived from an evaluation of various pre-selected voice samples over different transmission media, replayed under controlled conditions, to a statistically valid mixed group of men and women who assign a quality rating, or quality score, to the evaluated selection. Scores range from 1 to 5 and a score of 4.0 or higher is considered toll quality. A score of 1 is equated with bad (quality of speech—no meaning understood), a score of 2 is equated with poor (quality of speech—considerable effort), a score of 3 is equated with fair (quality of speech—moderate effort), a score of 4 is equated with good (quality of speech—attention necessary) and a score of 5 is equated with excellent (quality of speech—complete relaxation).
Objective tests, such as the perceptual speech quality measure (PSQM) and the perceptual evaluation of speech quality (PESQ), offer fast alternatives to subject testing. Although correlations between PESQ scores and ITU-T database subjective mean opinion score (MOS) results are around 93 percent, the objective metric does not provide a comprehensive evaluation of transmission quality. It only measures the effects of one-way speech distortion and noise on speech quality. The effects of loudness loss, delay, echo and other impairments related to two-way interaction are not reflected in the PESQ scores. On the other hand, these impairments are taken into consideration in the Emodel. As such, the coding system uses the R-Factor (R) as a measure of speech quality.
The Emodel is defined as follows:
Measured delay can be mapped into the effect of delay (Id) and packet loss into the effect of signal distortion (Ie). The effect of noise (Ro) and the account for quantization (IS) do not depend on network performance and are inherent to the voice signal itself. The R-Factor (R) can be translated into a mean opinion score (MOS) determination as set forth below:
For voice traffic, the R-Factor (R) is given the following equation:
The mapping from packet loss (Ie) is known for several types of speech codecs and packetization schemes. For the AMR speech codec, the effect of the signal distortion (Ie) versus frame erasure rate (FER) is shown on
As set forth above, the Emodel shows that increasing delay leads to reducing the MOS value, but decreasing packet loss leads to increasing the mean opinion score (MOS). Therefore, an optimum mean opinion score (MOS) value must exist for a given set of conditions. The coding system finds the optimum source-channel coding resulting in the best mean opinion score (MOS) score. As a next step, the coding system is capable of determining the end-to-end delay as well as the packet loss rate after forward error correction (FEC).
The end-to-end delay has various components. First, the encoding delay, which consists of the sum of the frame size T (20 ms for AMR) and the look-ahead delay Ia (which can consist of multiple processing delays at the transmission physical layer and the coder itself). Second, the packetization delay which accounts for the time it takes to group f frames together into one packet, i.e. Dpack=(f−1)T. The sum of the encoding delay and the packetization delay for k packets amounts to a formation time Tf=kTf+la. The processing delay for both the ACELP coder and the RS coder are assumed to be negligible.
The packets can then be transmitted on the access network 10, incurring transmission delay Th, queuing delay Qh, and propagation delay Ph at each hop h in the path from the source terminal 102, 106 to the destination terminal 120, 122. If Bh denotes the bandwidth at hop h, the total transmission delay is
In calculating the transmission delay (Tt) above, the present invention asumes that the first k speech packets can be transmitted as soon as they are formed provided that copies are kept in a buffer for the computation of the (n-k) FEC packets. The transmission of each of the first (k−1) packets occurs in parallel to the formation time of the next packet so it does not contribute to the total transmission time Tt. The transmission time for the kth packet does however contribute to Tt. In addition, (n-k) FEC packets are formed right after all of the k speech packets are generated and are transmitted in sequence, thus contributing to Tt (note that all n packets are of the same size under the FEC scheme set forth).
The propagation delay is negligible if within a local area; for intra-continental calls, the propagation delay is on the order of 30 ms and for inter-continental calls the delay can be as large as 100 ms. Upon reception, packets are usually delayed in a jitter buffer and the fixed playback delay must be at least the maximum queuing delay incurred in the network. The end-to-end delay can then be written as:
As an example, let us take k=2 and n=k, . . . , 10. Let us assume a 15-hop network path with one T1 line operating at 1.5 Mbps, 13 hops from the core network running at 622 Mbps, and one WLAN connection at 1 Mbps. Furthermore, let the queuing delay, propagation delay and playout buffer delay be zero. For each n/k value, the end-to-end delay (D) can then be computed and the corresponding effect of delay (Id) factor can be found. This example is illustrated in
The adaptive bit rate algorithm module 208 obtains estimates for packet loss rates on the wired and wireless hops, Pd and ps respectively, at the time an adaptation decision is being made. The QoS probing module 204 provides the packet loss rate on the wired hop (pd) as well as the end-to-end packet loss rate denoted by (ps), from which the packet loss rate on the wireless hop (ps) can be derived.
The packet loss measurement on the wireless hop is made after FEC schemes try to recover errors. Assuming a random loss model on the wireless hop, one can relate the measured packet loss on the wireless hop ps to the parameters of the FEC scheme, namely n and k, and the “raw” packet loss rate on the wireless link pr, i.e. what the packet loss would be in the absence of any correction scheme. Based on this information, the following equation can be written:
The data for Ie in
The preferred coding system 200 includes a bit rate adaptation algorithm module 208 that chooses appropriate source and channel bit rates given information on the available bandwidth and packet loss rates on the network path of interest. Given information on the available bandwidth (i.e., the maximum transmission rate that can be sustained by the network path such that congestion does not build up), and packet loss, the bit rate adaptation algorithm module 208 solves a bit allocation problem. The coding system 200 finds the optimum way of allocating source and channel bits such that the overall rate is less than the available bandwidth and the number of channel bits is adequate to result in desired speech quality given the packet loss rate (in effect, additional channel bits lead to a reduced packet loss since the code can recover from more errors). The tradeoff is as follows: increasing the number of channel bits makes the transmitted bits more robust to losses, but also decreases the goodput (i.e., the throughput of source bits) and increases the delay for source bits to arrive at the destination. The present coding system 200 analyzes this problem and finds the best possible combination of source and channel bit rates.
The source terminals 102, 106 may also include a source coding module 404 and a channel coding module 406 that are responsible for controlling the adaptive multi-rate coder 202 as dictated by the adaptive bit rate algorithm module 208. The destination terminals 120, 122 may include a channel decoder 408 and a source decoder 410 that are responsible for reproducing the speech signals that are contained in the continuous voice stream transmission from the source terminal 102, 106.
The inputs to the source and channel rate adaptation algorithm in the adaptive bit rate algorithm module 208 are preferentially: 1) QoS information: Wireless hop packet loss ps, path bandwidth (Bh) and congestion (Qh), and 2) QoS constraints: Maximum end-to-end delay Dmax, maximum allowed bandwidth AlBw, minimum goodput Gmin, and maximum permitted packet loss pmax. The output will be a decision on the choice of Rs and Rc, the source and channel bit rates respectively.
The adaptive bit rate algorithm module 208 determines the values of n and k such that the mean opinion score (MOS) is maximum, but subject to constraints on delay (Dmax), allowed bandwidth (AlBw), goodput (Gmin) and packet loss pmax.
The present invention starts by setting the following constraint:
Now the source code rate (Rs) can only take on values from among 8 possible rates: 4.75, 5.15, 5.90, 6.70, 7.40, 7.95, 10.2 and 12.2 kbps. It should be noted that 9 different rates are possible in wideband AMR. The bit rate adaptation algorithm module starts by picking the lowest value of 4.75 kbps for the source code rate (Rs). In the present invention, the values of k and n are varied over the ranges k=1, . . . , kmax and n=k, . . . , near. Note that n=k corresponds to the case without FEC. Parameter kmax is such that the delay (D) given by equation (4) is bounded by a maximum value (Dmax) beyond which voice quality becomes unacceptable (150-200 ms). Using equation (4) and equation (7):
Further, nmax is defined as:
For each (n, k) pair, the present invention obtains a value for D using equation (4) and this is mapped into a value for the effect of delay (Id) using the data from
At this stage, values for both the effect of delay (Id) and the effect of signal distortion (Ie) have been obtained, which correspond to a chosen tuple (n, k) and a source rate Rs. Once these values have been determined, the value of the R-Factor (R) may be obtained using equation (3), and then, the value of the mean-opinion-score (MOS) may be determined using equation (2). The process described above is repeated for all possible values of the source code rate (Rs). After this is accomplished, the bit rate adaptation algorithm module chooses the combination of the source code rate (Rs) and the channel code rate (Rc) that gives the maximum mean opinion score (MOS).
In another instance, a solution may not exist. In that case, the present invention does not use channel coding and picks the highest AMR mode that does not violate the allowed bandwidth constraint.
The bit rate adaptation algorithm module described above consists of an exhaustive search over a relatively small set of values. The number of modes for the AMR codec is typically 8 (9 for wideband AMR) and n and k are typically less than 20 or 25 at most. This means the algorithm requires relatively few computational operations to complete making it suitable for real-time and dynamic adaptation as the voice over IP communication progresses.
In the following discussion, the following assumptions are made about the setup of the IP access network 100 and voice environment:
The plots in
The result in
As set forth in detail above, the present invention discloses an adaptation algorithm of source and channel coding rates to QoS conditions in wireless IP networks using the AMR speech codec and Reed-Solomon codes. The metric used for optimization is the mean opinion score (MOS) as defined by the ITU-T Emodel for voice traffic. The present invention shows that for noisy links and given QoS constraints it is necessary to sacrifice source bits for increased robustness to packet loss. Moreover, the present invention acknowledges the fact that while FEC mitigates the effects of packet loss, it also increases the end-to-end delay. These two trends work opposite to each other, the first causing an increase and the second a decrease in voice quality. The algorithm finds the optimum compromise between packet loss recovery and end-to-end delay to maximize perceived voice quality.
While various embodiments of the invention have been described, it will be apparent to those of ordinary skill in the art that many more embodiments and implementations are possible within the scope of the invention. Accordingly, the invention is not to be restricted except in light of the attached claims and their equivalents.
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